VoS on Discord!

Now that I’ve set up a brand new Variety of Sound Discord server, you are of course all welcome to join in. Naturally, the main focus will be on the plugins and everything related to them, but I’m also looking forward to getting to know some of you a bit better.

I hope everything is working so far, but if you have any problems logging in, just let me know here with a short message – thanks in advance! Thanks also to rktic for supporting me setting up the server!

The upcoming public beta will be available on the server in the next few days and I’m looking forward to many downloads and diligent testing and feedback, as always 🙂

Click here to join the VoS Discord server. A Discord login is required. Once on my server, you must agree the welcome message just shown to you by ticking it with a “thumbs up” to get full rights for posting etc.

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VoS on Discord?

What will be the next plugin release from “Variety Of Sound”?

I just asked this to ChatGPT and here is the answer:

As an AI language model, I don’t have access to real-time information about upcoming releases from specific companies or developers. However, it’s worth noting that “Variety Of Sound” is a well-known developer of high-quality audio plugins, with a particular focus on analog-style signal processing.

In the past, “Variety Of Sound” has released a range of plugins, including EQs, compressors, tape saturation emulators, and more. These plugins are known for their attention to detail, their intuitive interfaces, and their ability to add warmth and character to digital audio signals.

While I don’t have information about any specific upcoming releases from “Variety Of Sound”, it’s possible that they may continue to expand their range of analog-style plugins, or explore new areas of audio processing. As with any developer, it’s likely that they will release new products based on their own interests and expertise, as well as feedback and requests from their user base.

So, what about “feedback and requests” from your side?

why the Thrillseeker compressors complement each other so well

Audio compressors use either a “feed forward” or “feedback” design to control the gain of an audio signal. In a feed forward compressor, the input signal is used directly to control the gain of the output signal. Essentially, the compressor compares the input signal to a threshold and reduces the gain of the output signal if the input signal exceeds the threshold. In a feedback compressor, the output signal is fed back into the compressor and used to control the gain of the input signal. So, the compressor compares the output signal to a threshold and reduces the gain of the input signal if the output signal exceeds the threshold. Both feed forward and feedback compressors can be effective at controlling the dynamic range of an audio signal, but they operate in slightly different ways and do have different characteristics in terms of their sound and response.

However, the specific sound of a device depends largely on other features of the circuit design and its components. For example, an optoelectric compressor uses a photoresistor or photodiode to detect and control the degree of gain reduction of the signal. But the make-up amplifier afterwards may contribute the most to the sound, depending on its design (tube or solid state). A variable gain tube compressor, on the other hand, uses a vacuum tube to control the gain of the compressor. The vacuum tube is used to amplify the signal, and the gain of the compressor is controlled by changing the bias voltage of the tube. This alone provides a very typical, distinctive sound that is very rich in harmonic overtones.

Both opto-electrical and variable-mu tube compressors are commonly used in audio production to control the dynamic range of a signal, but they operate in different ways and can produce different tonal characteristics. Opto-electrical compressors are known for their fast attack times and smooth release characteristics, while variable-mu tube compressors are known for their warm and smooth sound.

interview series (11) – Andreas Eschenwecker

Andy, your Vertigo VSC compressor has already become a modern classic. What has been driven you to create such a device?

I really like VCA compressors. VCA technology gives you a lot of freedom in design and development and the user gets a very flexible tool at the end. I was very unhappy with all VCA compressors on the market around 2000. Those were not very flexible for different applications. These units were working good in one certain setting only. Changing threshold or other parameters was fiddley and so on. But the main point starting the VSC project was the new IC VCA based compressors sounded one dimensional and boxy.

Does this mean your design goal was to have a more transparent sounding device or does the VSC also adds a certain sound but just in a different/better way?

Transparency without sounding clean and artificial. The discrete Vertigo VCAs deliver up to 0,6% THD. Distortion can deliver depth without sounding muddy.

Does this design favour certain harmonics or – the other way around – supresses some unwanted distortions?

The VSC adds a different distortion spectrum depending when increasing input level or adding make-up. The most interesting fact is that most of the distortion and artifacts are created in the release phase of the compressor. The distortion is not created on signal peaks. It’s becoming obvious when the compressor sets back from gainreduction to zero gainreduction. Similar to a reverb swoosh… after the peak that was leveled.

Where does your inspiration comes from for such technical designs?

With my former company I repaired and did measurements on many common classic and sometimes ultra-rare compressors. Some sounded pretty good but were unreliable – some were very intuitive in a studio situation, some not…
At this time I slowly developed an idea what kind of compressor I would like to use in daily use.

From your point of view: To which extend did the compressor design principles changed over the years?

The designs changed a lot. Less discrete parts, less opto compressors (because a lot of essential parts are no longer produced), tube compressors suffer from poor new tube manufacturing and some designers nowadays go more for RMS detection and feed forward topology. With modern components there was no need for a feedback SC arrangement anymore. I think RMS is very common now because of its easy use at the first glance. For most applications I prefer Peak detection.

Having also a VSC software version available: Was it difficult to transfer all that analog experience into the digital domain? What was the challenge?

In my opinion the challenge is to sort out where to focus on. What influence has the input transformer or the output stage? Yes some of course. Indeed most of the work was going into emulating the detection circuit.

Which advantages did you experienced with the digital implementation or do you consider analog to be superior in general?

I am more an analog guy. So I still prefer the hardware. What I like about the digital emulations is that some functions are easy to implement in digital and would cost a fortune in production of the analog unit.

Any plans for the future you might want to share?

At the moment I struggle with component delays. 2021/22 is not the right time for new analog developments. I guess some new digital products come first.

Related Links

• Vertigo Sound

interview series (10) – Vladislav Goncharov

Vlad, what was your very first DSP plugin development, how did it once started and what was your motivation behind?

My first plugin was simple a audio clipper. But I decided to not release it. So my first public released plugin was Molot compressor. I was a professional software engineer but with zero DSP knowledge (my education was about databases, computer networks and stuff like that). I played a guitar as a hobby, recorded demos at home and one day I found that such thing as audio plugins exist. I was amazed by their amount and also by the fact that there are free plugins too. And I realised that one day I can build something like this myself. I just had to open a DSP book, read a chapter or two and it was enough to start. So my main motivation was curiosity, actually.

Was that Molot compressor concept inspired by some existing devices or a rather plain DSP text book approach?

That days there was a rumour that it’s impossible to make good sounding digital compressor because of aliasing and stuff. I tried to make digital implementation as fluid as possible, without hard yes/no logic believing this is how perfect digital compressor should sound. And the way I implemented the algorithm made the compressor to sound unlike anything I heard before. I didn’t had any existing devices in my head to match and I didn’t watch textbook implementations too. The sound was just how I made it. I did 8 versions of the algorithm trying to make it as usable as possible from user point perspective (for example “harder” knee should sound “harder”, I removed dual-band implementation because it was hard to operate) and the last version of the project was named “comp8”.

Did you maintained that specific sound within Molot when you relaunched it under the TDR joint venture later on? And while we are at it: When and how did that cooperation with Fabien started?

TDR Molot development was started with the same core sound implementation as original Molot had. But next I tried to rework every aspect of the DSP to make it sound better but keep the original feel at the same time. It was very hard but I think I succeeded. I’m very proud of how I integrated feedback mode into TDR Molot for example. About Fabien: He wrote me to discuss faults in my implementation he thought I had (I’m not sure it was Molot or Limiter 6), we also discussed TDR Feedback Compressor he released that days, we argued against each other but what’s strange the next day we both changed our minds and agreed with our opposite opinions. It was like “You were right yesterday. No, I think you were right”. Next there was “KVR Developer Challenge” and Fabien suggested to collaborate and create a product for this competition. That was 2012.

And the Feedback Compressor was the basis for Kotelnikov later on, right?

No, Kotelnikov is 100% different from Feedback Compressor. Fabien tried to make the sound of feedback compressor more controllable and found that the best way to achieve this is just to change the topology to feedforward one. It’s better to say, Feedback Compressor led to Kotelnikov. Also the interesting fact, early version of Kotelnikov had also additional feedback mode but I asked Fabien to remove it because it was the most boring compressor sound I ever heard. I mean if you add more control into feedback circuit, it just ruins the sound.

Must have been a challenge to obtain such a smooth sound from a feed-forward topology. In general, what do you think makes a dynamic processor stand out these days especially but not limited to mastering?

I think, it’s an intelligent control over reactions. For example Kotelnikov has some hidden mechanisms working under the hood, users don’t have access to them but they help to achieve good sound. I don’t think it’s good idea to expose all internal parameters to the user. There must be hidden helpers just doing their job.

I so much agree on that! Do you see any new and specific demand concerning limiting and maximizing purposes? I’m just wondering how the loudness race will continue and if we ever going to see a retro trend towards a more relaxed sound again …

I think even in perfect loudness normalized world most of the music is still consumed in noisy environments. The processing allowing the quietest details to be heard and cut through background noise, to retain the feel of punch and density even at low volumes is in demand these days. Loudness maximizers can do all this stuff but in this context they act like old broadcast processors. In my opinion, the loudness war will continue but it’s not for overall mix loudness anymore but how loud and clear each tiny detail of the mix should be.

Can we have a brief glimpse on what you are currently focused on, DSP development wise?

You may take a look at Tokyo Dawn Labs Facebook posts. We shared a couple of screenshots some time ago. That’s our main project to be released someday. But also we work on a couple of dynamic processors in parallel. We set high mark on the quality of our products so we have to keep it that high and that’s why the development is so slow. We develop for months and months until the product is good enough to be released. That’s why we usually don’t have estimation dates of release.

Related Links

• Tokyo Dawn Labs

the many shapes of delay

About the different applications of audio delay effects.

There are quite a number of different types and applications for the audio delay effect in the audio production ranging from plain technical delay application up to all the musical and creative ranges of application. The rather technical or correctional delay typically is a plain digital delay which serves as a sample accurate alignment tool. Such alignments might be necessary for example for plug-in delay compensation (when a plug-in introduces latency) or during the mixing process to align a group of recorded tracks or samples. The so-called pre-delay can also be seen as an alignment sort of thing where the direct source signal has to be aligned in a positive or negative manner in relation to a processed signal, e.g. in a reverb effect.

I do believe in feedback

If I would have to state just one single universal principle which life as a whole rests upon then I would probably refer to the principle of feedback. But don’t worry, I don’t want to indulge in rather esoteric directions here but just move on a little bit towards basic system theory and how this relates to audio processing.

(source: wikipedia.org)

According to classical control theory, such systems are taking its output information and feeding it back in to the process input and that way closing the loop – hence the name closed loop system. In DSP audio land, the information is the audio signal itself and the audio (feedback) path constitutes the closed loop system. Audio signal processors such as feedback compressors or guitar amp effects are good examples for specific applications. While the control theory provides a lot of guidance for closed loop systems e.g. on handling differential equations and stability criteria, this can get pretty much nasty in practise because of the potential manifold of feedback loops in the circuit and not just only the main audio path.

NastyDLA – final teaser and release info

NastyDLA – technical architecture

simplified technical architecture

Internally, NastyDLA consists of quite a bunch of DSP processing building blocks which as a whole are summing up to an authentic signal path simulation of it’s analog models. The blocks and the according signal flow are shown in the diagram above. Basic signal flow goes from left to right except the feedback path which goes in the opposite direction.

With NastyDLA, signal path coloration already starts in the input stage which provides a complete model of both, frequency and phase response as well as dynamic saturation. It’s located in the dry path but all nonlinear processing and coloring can be disabled on demand so it remains as a simple input volume control then. But while switched in, the input stage can greatly contribute on getting the processed signal to fit right into a mix.

NastyDLA – some tape delay fun

To make up a complete and sustaining sequence out of  some melodic pattern is a standard task for the electronic musician. The good old tape delay is his best friend then, providing not only consistent and sustainable echos which are glueing nicely with the original signal but also offering some realtime modulation possibilities as well, to animate some maybe rather static sources.

In this short demo a static pattern is used and NastyDLA is going to be in charge as a tape delay replacement. The chorus is not used in this example but the plug-ins coloring possibilities are shown to some extend: After some bars the timbre of the delay feedback loop changes to a higher pitch and then to a lower one (and vice versa) while simultaneously the feedback amounts are going to change. To the end when the pattern stops, the “tape speed” is slowed down first and accelerated back again afterwards to demonstrate its artifact-free modulation capabilities.

Note how smooth the saturation behaves when driven into self oscillation w/o the usual amount of aliasing artifacts. The delay line generation in this example is set to “dual mono” mode (with 8th to the left and dotted 8th to the right) and the time modulations can be done separate per channel. All animations were done in realtime with host automation and in general all plug-in parameters can be automated in the host.